route in each audio stream matching the routes used for each SIP account you have
registered to the codec. In this way, when a call is received via a particular account, the call
is routed to the audio stream with the matching audio answer route. This process also
reliably predetermines which inputs and outputs are used on the answering codec (e.g.
Input 1 and Output 1 for the first audio stream in a multiple stream program).
2. SIP Interfaces: When answering multiple incoming peer-to-peer SIP calls, configure an
answer Route for incoming peer-to-peer calls via the SIP Interface panel and route them to
specific audio streams. Configure answer routes in Interface SIP1 and Interface SIP2 to
match answer routes configured in an answering program's audio streams. This process also
reliably predetermines which inputs and outputs are used on the answering codec (e.g.
Input 2 and Output 2 for the second audio stream in a multiple stream program).
Important Notes:
© Tieline Pty. Ltd. 2017
·
Ports are arbitrarily assigned when answering calls using SIP accounts. By default
the UDP audio port range used is from 5004 to 5054. This setting can be adjusted in
the SIP Interfaces panel in the HTML5 Toolbox web-GUI. Ensure your firewall has
the required TCP and UDP ports open if you are receiving multiple SIP calls.
·
Remember to lock an answering program in a codec when answering multiple SIP
calls.
·
Ensure the appropriate TCP and UDP audio ports are open in your firewall to allow
multiple SIP audio streams to connect. See
more information.
Merlin PLUS WNET Manual v2.0
Installing the Codec at the Studio
225
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